CN101341793A - Method to generate multi-channel audio signals from stereo signals - Google Patents

Abstract

A perceptually motivated spatial decomposition for two-channel stereo audio signals, capturing the information about the virtual sound stage, is proposed. The spatial decomposition allows to re-synthesize audio signals for playback over other sound systems than two-channel stereo. With the use of more front loudspeakers, the width of the virtual sound stage can be increased beyond +/- 30 DEG and the sweet spot region is extended. Optionally, lateral independent sound components can be played back separately over loudspeakers on the two sides of a listener to increase listener envelopment. It is also explained how the spatial decomposition can be used with surround sound and wavefield synthesis based audio system. According to the main embodiment of the invention applying to multiple audio signals, it is proposed to generate multiple output audio signals (y1,..., yM) from multiple input audio signals (x1, ..., xL), in which the number of output is equal or higher than the number of input signals , this method comprising the steps of: by means of linear combinations of the input subbands (X1(i), ..., XL(i)), computing one or more independent sound subbands representing signal components which are independent between the input subbands, by means of linear combinations of the input subbands (X1(i), ..., XL(i)), computing one or more localized direct sound subbands representing signal components which are contained in more than one of the input subbands and direction factors representing the ratios with which these signal components are contained in two or more input subbands, generating the output subband signals (Y1(i)...YM(i)), where each output subband signal is a linear combination of the independent sound subbands and the localized direct sound subbands, converting the output subband signals (Y1(i)...YM(i)), to time domain audio signals (y1,..., yM).

Description

Produce the method for multi-channel audio signal from stereophonic signal

Background technology

Many technological innovations beyond the stereophony are because cost, can't implementation (for example, the number of loud speaker) fail, and last but be not the demand that least importantly is used for back compatible.And 5.1 adopted by the user widely around the multichannel audio system, in addition, this system is with regard to the number of loud speaker, and since the restriction of back compatible (a preceding left side is placed on the angle identical with stereophony with right loud speaker, promptly+/-30 °, cause narrow positive virtual sound level) be half measure.

The fact further most of audio contents in two-channel stereo format are available.For the audio system that strengthens stereo sound experience in addition, compare with legacy system, can play the stereo audio content by means of the experience that improves therefore be crucial yearningly.

Realized using more preceding loud speaker to improve virtual sound level for a long time for the listener in the central point that not exclusively is arranged on club face in addition.For improving the purpose of result's existence through plural loudspeaker plays stereophonic signal.Especially, there are many concerns for utilizing extra center loudspeaker to play stereophonic signal.But the improvement of these technology on the played in stereo of routine is not clear enough, and they are widely used.The major limitation of these technology is that they only consider the position, and do not consider other aspect clearly, such as surrounding environment and listener's envelope.In addition, the situation theory after these technology is based on a virtual information source situation, also limits its performance when many information sources side by side come across different directions.

But these weakness are to excite the technology of the spatial decomposition of stereo audio signal to overcome by what propose in this manual with passing through the use perception.Provide this and decompose, can present audio signal for the capable array of loud speaker, loud speaker and the wave field synthesis system that increase number.

The technology of this proposition is for by means of more sound channel stereophonic signal (two sound channels) being converted to audio signal without limits.But the signal that normally, has L sound channel can be converted into the signal with M sound channel.This signal can or stereo, or purpose is the multi-channel audio signal that is used to play, perhaps they can be unprocessed microphone signals, perhaps the linear combination of microphone signal.It also illustrates this technology and how to be applied to microphone signal (for example, ambiophony sound B form), and matrix be used in various loud speaker general layouts, reproducing these around following mixed frequency signal.

When we mentioned stereo or have the multi-channel audio signal of many sound channels, we refer to when we mention many (monophone) audio signals was identical.

Summary of the invention

According to the main embodiment that is applied to a plurality of audio signals, it has proposed from a plurality of input audio signal (x ₁..., x _L) a plurality of output audio signal (y of middle generation ₁..., y _M), wherein Shu Chu number equals or is higher than the number of input signal, and this method comprises step:

-utilize and import sub-band X ₁(i) ..., X _LThe mode of linear combination (i) is calculated one or more independently sound sub-bands of representing signal component, and this signal component is independently between the input sub-band;

-utilize and import sub-band X ₁(i) ..., X _LThe mode of linear combination (i), calculate the direct sound wave sub-band of one or more parts of expression signal component, this signal component be comprised in the input sub-band more than one in, with the direction factor of expression ratio, these signal components are included in this ratio in two or more input sub-bands;

-produce and export sub-band signal Y ₁(i) ... Y _M(i), here each output sub-band signal is the independently linear combination of sound sub-band and local direct sound wave sub-band;

-will export sub-band signal Y ₁(i) ... Y _M(i) be converted to time-domain audio signal y ₁... y _M

This index i is the index of the sub-band of consideration.According to first embodiment, this method can each audio track a sub-frequency bands and using only, even the more sub-band of each sound channel provides the better sound result.

The scheme of this proposition is based on following reason.Many input audio signal x ₁..., x _LBe broken down into the signal component of expression sound, this sound be between audio track and signal component independently, this signal component is illustrated between the audio track relevant sound.This different consciousness effect that is the signal component by these two types has inspires.This independently signal component represent the information of relevant information source width, listener's envelope and surrounding environment, and should relevant (subordinate) signal component represent the position or the sense of hearing ground direct sound wave of auditory events.For each relevant signal component, there is relevant directional information, it can be by one than value representation, and this sound is included in many audio input signals with this ratio.When loud speaker (perhaps headphone) is gone up broadcast,, the purpose of reproducing specific auditory space image decomposes the many audio output signals of generation for can providing this.Should relevant signal component be rendered as output signal (y ₁..., y _M), make it by the directional perception of listener from expectation.This independently signal component be rendered as output signal (loud speaker), make it simulate the consciousness effect of non-direct sound wave and its expectation.This function of describing on high standard is to extract spatial information from input audio signal, and this spatial information is transformed to the spatial information that has the parameter of expectation in this output channels.

Description of drawings

Because additional accompanying drawing will be understood the present invention better, wherein:

Fig. 1 illustrates the setting of standard boombox;

Fig. 2 illustrates the position for the auditory events of the perception of the different level difference value of two relevant loudspeaker signals, determines the position of the auditory events between present two loud speakers at the level between a pair of relevant loudspeaker signal and time difference;

Fig. 3 (a) illustrates the early reflection of sending from the side loud speaker with auditory events expansion effect;

Fig. 3 (b) illustrates the later stage reflection of sending from the side loud speaker that relates to as more environment of listener's envelope;

Fig. 4 illustrates stereophonic signal and side-reflected mode of hybrid analog-digital simulation direct sound wave;

It is the T/F tiling demonstration of sub-band with signal decomposition that Fig. 5 illustrates expression as the function of time;

Fig. 6 illustrates the normalization power of direction factor A and S and AS;

Fig. 7 illustrates least square estimation weight w ₁And W ₂, and the back scale factor that is used to calculate estimation s;

Fig. 8 illustrates least square estimation weight w ₃And w ₄, and be used for calculating estimation N ₁Back scale factor;

Fig. 9 illustrates least square estimation weight w ₅And w ₆, and be used for calculating estimation N ₂Back scale factor;

Figure 10 illustrates s, A, the n of estimation ₁And n ₂

Figure 11 illustrates ± 30 ° of virtual sound levels (a) is converted to the have loudspeaker array virtual sound level of width in slit of (b);

Figure 12 illustrate loud speaker to select 1 with the factor a relevant with the stereophonic signal level difference ₁And a ₂

Figure 13 illustrates the plane wave that sends via a plurality of loud speakers;

Figure 14 illustrates the virtual sound level that ± 30 ° of virtual sound levels (a) is converted to the width in the slit with loudspeaker array, and sound improves listener's envelope by sending independently from side loudspeaker (b);

Figure 15 illustrate for as eight signals that generation is set in Figure 14 (b);

Figure 16 illustrates each signal corresponding to the preceding sound level that is interpreted as virtual source.This independently horizontal sound is used as plane wave (virtual source in the far field) and sends;

Figure 17 illustrates quadraphonic sound system (a) and expands to for more loud speaker (b) and use.

Embodiment

The space is listened attentively to boombox and is play

The scheme of this proposition inspires the description for the important condition of two input sound channels (stereo audio input) and M audio frequency output channels (M 〉=2).After a while, how its description will be applied to the situation of more conventional L input sound channel with the identical reason of deriving in the example of stereo input signal.

The most normally used user's Play System that is used for space audio is boombox setting as shown in Figure 1.Two loud speakers are placed on this listener's left side and front, right side.Usually, these loud speakers are arranged on the circle with angle-30 ° and+30 °.The width of the auditory space image of perception when listening to such stereo playing system is approximate to be limited between two loud speakers and the zone after two loud speakers.

When listening to naturally and working as the sound of listening to reproduction, the auditory space image of this perception mainly depends on the ears position indicating, that is, and and coherence (IC) between level difference (ILD) and ear between interaural difference (ITD), ear.In addition, its elevation angle that perception has been shown is relevant with monaural prompting.

Make that the ability of playing the auditory space image that generates the simulation sound level by means of boombox is possible by the consciousness phenomenon of position summation, promptly, by being controlled at level and/or the time difference between the signal of giving this loud speaker, auditory events can with the loud speaker of listener front between any angle occur.In nineteen thirty, Blumlein recognizes the power of this principle, and the present famous relevant stereophonic patent of his application.The position summation is based on the following fact, that is, the approximate crudely dominant prompting of ITD that on ear, causes and ILD prompting, if physical resource is located on the direction of the auditory events that occurs between the loud speaker, it will occur.

Fig. 2 illustrates the position for the auditory events of the different level difference perception of two relevant loudspeaker signals.When a left side and right loudspeaker signal are concerned with, have identical level, and when postpone differing from, an auditory events appears at two central authorities between the loud speaker, as illustrational by the zone among Fig. 21.By in a side, for example improve level on the right side, this auditory events moves to as by 2 illustrational those sides of the zone in Fig. 2.Under unusual situation, when only the signal on the left side is effective, appear at locational this auditory events of left speaker as illustrational by the zone in Fig. 23.Can control the position of this auditory events similarly by the delay of change between loudspeaker signal.When this loud speaker in the front not the listener, be controlled at loud speaker between the described principle in auditory events position also be applicatory.But, for some restricted applications of loud speaker in listener's side.

As in Fig. 2 illustrated, the position summation can be used for simulating a kind of situation, and different here instruments is being positioned on the virtual sound level on the different directions, that is, and and in the zone between two loud speakers.Hereinafter, except can the control position, how description can control other attribute.

As one man important venue acoustics is to consider the reflection that arrives on the listener from the side, that is, and and sideswipe.Original sideswipe has been shown has had the effect that enlarges auditory events.It is constant having effect less than the approximately primary reflection of the delay of 80ms approximate, and therefore, has been defined in the concrete measure of considering the lateral part that primary reflection is represented in this scope.This lateral part is the ratio of horizontal acoustic energy to total acoustic energy, and total acoustic energy is after direct sound wave arrives at, and obtains in initial 80ms, and measures the width of auditory events.

Be used to imitate early stage side-reflected experimental facilities in Fig. 3 (a) illustrated.This direct sound wave sends from center loudspeaker, and independently early reflection is sent from left side and right speaker.When early stage side-reflected relative intensity improved, the width of this auditory events increased.

After direct sound wave arrived at, the above sideswipe of 80ms tended to help more the perception of environment except that auditory events itself.This is tangible on the meaning of " envelope " of often representing listener's envelope or " broad environment ".Also measure the reflection in the later stage of listener's envelope degree applicable to confession as the similar measure of the lateral part that is used for early reflection.This measures the transverse energy part in expression later stage.Can be with the sideswipe that the simulation later stage is set shown in Fig. 3 (b).This direct sound wave sends from center loudspeaker, and independently the reflection in later stage is sent from left side and right speaker.When the side-reflected relative intensity in later stage improved, the sensing of this listener's envelope increased, simultaneously the width of this auditory events be expect affected hardly.

Stereophonic signal is recorded or mixes, make for each information source, this signal enters left side and right-side signal sound channel with specific direction prompting (level difference, time difference) consistently, and the independently signal of reflection/repercussion enters the sound channel of determining auditory events width and the prompting of listener's envelope.Further discussing mixing and recording technique is beyond the scope of this specification.

The spatial decomposition of stereophonic signal

With use on the contrary from the direct sound wave of true information source, as in Fig. 3 illustrated, people can use the direct sound wave corresponding to the virtual source of utilizing the position summation to produce.The auditory events of perception is represented in this shadow region.That is to say, can only realize by means of two loud speakers as experiment shown in Figure 3.These are in Fig. 4 illustrated, and signal s simulation here comes the direct sound wave of the definite direction of free factor a.This independently signal n1 and n2 corresponding to sideswipe.The situation of this description is by means of the natural decomposition of an auditory events for stereophonic signal,

x ₁(n)＝s(n)+n ₁(n) x ₂(n)＝as(n)+n ₂(n)

(1)

Catch the position and the width of this auditory events and listener's envelope.

In order to decompose, under an auditory events situation, it is not only effectively, but, having the non-static case of a plurality of effective information sources simultaneously, the decomposition of this description is independently at many frequency band ranges with realize in the time adaptively,

X ₁(i，k)＝S(i，k)+N ₁(i，k) X ₂(i，k)＝A(i，k)S(i，k)+N ₂(i，k)

(2)

Here i is the sub-band index, and k is the sub-band time index.This promptly, shows this signal S, N in each the T/F tiling with index i and k in Fig. 5 illustrated ₁, N ₂Estimated independently with direction factor A.For mark for simplicity, this sub-band and time index are left in the basket hereinafter usually.We use sub-band to decompose by means of consciousness ground exciton band bandwidth, that is, the bandwidth of sub-band is selected to equal a critical band.About every 20ms estimation S, N in each sub-band ₁, N ₂With direction factor A.

Notice that in general, people also can consider the time difference of direct sound wave in formula (2).That is to say that people are the service orientation factors A not only, and the service orientation delay, this direction postpones to be defined as having S and is included in X ₁And X ₂In delay.In the following description, we do not consider above-mentioned delay, still, should be understood that this analysis can easily expand to the above-mentioned delay of consideration.

Provide stereo sub-band signal X ₁And X ₂, this target is to calculate S, N ₁, N ₂Estimated value with A.X ₁The estimated value in short-term of power be expressed $P_{x1}(i,k)=E\left\{{\mathrm{<figure-callout\; id="1"\; label="X"\; filenames="A20068003222800292.png,A20068003222800301.png"\; state="\{\{state\}\}">X</figure-callout>}}_1^2(i,k)\right\}.$ For other signal, use identical agreement, that is, and P _X2, Ps and P _N=P _N1=P _N2It is power estimated value in short-term accordingly.N ₁And N ₂Power to be assumed to be identical, that is, suppose that laterally independently amount of sound is the same to a left side with the right side.

Notice, can use except that P _N=P _N1=P _N2Outside other hypothesis.For example, A ²P _N1=P _N2

Estimation Ps, A and P _N

The sub-band that provides this stereophonic signal is represented, calculates this power (P _X1, P _X2) and standardized cross-correlation.Standardized cross-correlation between a left side and the right side is:

$\mathrm{\&Phi;}(i,k)=\frac{E\left\{X_1(i,k)X_2(i,k)\right\}}{\sqrt{E\left\{{\mathrm{<figure-callout\; id="1"\; label="X"\; filenames="A20068003222800292.png,A20068003222800301.png"\; state="\{\{state\}\}">X</figure-callout>}}_1^2(i,k)\right\}E\left\{{\mathrm{<figure-callout\; id="2"\; label="X"\; filenames="A20068003222800282.png,A20068003222800311.png"\; state="\{\{state\}\}">X</figure-callout>}}_2^2(i,k)\right\}}}---\left(3\right)$

A, Ps and P _NBe calculated as the P of estimation _X1, P _X2Function with Ф.Relate to the sixth of the twelve Earthly Branches and know and three formula of unknown variable are:

Px ₁＝P _S+P _N Px ₂＝A ²P _S+P _N $\mathrm{\&Phi;}=\frac{\mathrm{aS}}{\sqrt{{\mathrm{<figure-callout\; id="1"\; label="Px"\; filenames="A20068003222800292.png,A20068003222800301.png"\; state="\{\{state\}\}">Px</figure-callout>}}_1}{\mathrm{Px}}_2}---\left(4\right)$

These formula are obtained A, P _SAnd P _N, obtain:

$A=\frac{B}{2C}$ $P_S=\frac{{2C}^2}{B}$ $P_N=X_1-\frac{{2C}^2}{B}---\left(5\right)$

And

$B={\mathrm{Px}}_2-{\mathrm{<figure-callout\; id="1"\; label="Px"\; filenames="A20068003222800292.png,A20068003222800301.png"\; state="\{\{state\}\}">Px</figure-callout>}}_1+\sqrt{{({\mathrm{Px}}_1-{\mathrm{Px}}_2)}^2+4{\mathrm{<figure-callout\; id="1"\; label="Px"\; filenames="A20068003222800292.png,A20068003222800301.png"\; state="\{\{state\}\}">Px</figure-callout>}}_1{\mathrm{<figure-callout\; id="2"\; label="Px"\; filenames="A20068003222800282.png,A20068003222800311.png"\; state="\{\{state\}\}">Px</figure-callout>}}_2{\mathrm{\&Phi;}}^2}$ $C=\mathrm{\&Phi;}\sqrt{{\mathrm{<figure-callout\; id="1"\; label="Px"\; filenames="A20068003222800292.png,A20068003222800301.png"\; state="\{\{state\}\}">Px</figure-callout>}}_1{\mathrm{Px}}_2}---\left(6\right)$

S, N ₁And N ₂Least square estimation.

Next, S, N ₁And N ₂Least square estimation be calculated as A, Ps and P _NFunction.For each i and k, this signal S is estimated as:

$\hat{S}={\mathrm{\&omega;}}_1{\mathrm{<figure-callout\; id="1"\; label="X"\; filenames="A20068003222800292.png,A20068003222800301.png"\; state="\{\{state\}\}">X</figure-callout>}}_1+{\mathrm{\&omega;}}_2{X}_2={\mathrm{\&omega;}}_1(S+N_1)+{\mathrm{\&omega;}}_2(\mathrm{AS}+N_2)---\left(7\right)$

Here ω ₁And ω ₂It is real-valued weight.This estimation error is:

E＝(1-ω ₁-ω ₂A)S-ω ₁N ₁-ω ₂N ₂ (8)

When this error E is to be orthogonal to X ₁And X ₂The time, this weights omega ₁And ω ₂In the lowest mean square sensing, be best, that is,

E{EX ₁}＝0 E{EX ₂}＝0 (9)

Obtain two formula,

(1-ω ₁-ω ₂A)P _S-ω ₁P _N＝0，

A(1-ω ₁-ω ₂A)P _S-ω ₂P _N＝0 (10)

This is extremely heavily from wherein being calculated,

${\mathrm{\&omega;}}_1=\frac{P_S{P}_N}{(A^2+1)P_S{P}_N+{\mathrm{<figure-callout\; id="2"\; label="P\; N"\; filenames="A20068003222800282.png,A20068003222800311.png"\; state="\{\{state\}\}">P</figure-callout>}}_N^{}}$ ${\mathrm{\&omega;}}_2=\frac{{\mathrm{AP}}_S{P}_N}{(A^2+1)P_S{P}_N+P_N^2}---\left(11\right)$

Similarly, N ₁And N ₂Estimated.N ₁Estimated value be:

${\hat{N}}_1={\mathrm{\&omega;}}_3{\mathrm{<figure-callout\; id="1"\; label="X"\; filenames="A20068003222800292.png,A20068003222800301.png"\; state="\{\{state\}\}">X</figure-callout>}}_1+{\mathrm{\&omega;}}_4{X}_2={\mathrm{\&omega;}}_3(S+N_1)+{\mathrm{\&omega;}}_4(\mathrm{AS}+N_2)---\left(12\right)$

This estimation error is:

E＝(-ω ₃-ω ₄A)S-(1-ω ₃)N ₁-ω ₂N ₂ (13)

Equally, calculating this weight makes this estimation error be orthogonal to X ₁And X ₂, the result forms:

${\mathrm{\&omega;}}_3=\frac{A^2{P}_S{P}_N+P_N^2}{(A^2+1)P_S{P}_N+{\mathrm{<figure-callout\; id="2"\; label="P\; N"\; filenames="A20068003222800282.png,A20068003222800311.png"\; state="\{\{state\}\}">P</figure-callout>}}_N^{}}$ ${\mathrm{\&omega;}}_4=\frac{{-\mathrm{AP}}_S{P}_N}{(A^2+1)P_S{P}_N+P_N^2}---\left(14\right)$

Be used to calculate N ₂The weight of least square estimation be:

${\hat{N}}_2={\mathrm{\&omega;}}_5{\mathrm{<figure-callout\; id="1"\; label="X"\; filenames="A20068003222800292.png,A20068003222800301.png"\; state="\{\{state\}\}">X</figure-callout>}}_1+{\mathrm{\&omega;}}_6{X}_2={\mathrm{\&omega;}}_5(S+N_1)+{\mathrm{\&omega;}}_6(\mathrm{AS}+N_2)---\left(15\right)$

Be

${\mathrm{\&omega;}}_5=\frac{-A{P}_S{P}_N}{(A^2+1)P_S{P}_N+{\mathrm{<figure-callout\; id="2"\; label="P\; N"\; filenames="A20068003222800282.png,A20068003222800311.png"\; state="\{\{state\}\}">P</figure-callout>}}_N^{}}$ ${\mathrm{\&omega;}}_6=\frac{P_S{P}_N+P_N^2}{(A^2+1)P_S{P}_N+P_N^2}---\left(16\right)$

Back scale

Provide the least square estimation, these are made and estimate by (selectively) back scale

Power equal Ps and PN=P _N1=P _N2

Power be:

$P_{\stackrel{\&CenterDot;}{S}}={({\mathrm{\&omega;}}_1+a{\mathrm{\&omega;}}_2)}^2{P}_S+({\mathrm{\&omega;}}_1^2+{\mathrm{\&omega;}}_2^2)P_N---\left(17\right)$

Therefore, for by means of by the power Ps of scale,

Obtain the estimated value of S:

${\hat{S}}^{\&prime;}=\frac{\sqrt{P_N}}{\sqrt{{({\mathrm{\&omega;}}_1+a{\mathrm{\&omega;}}_2)}^2{P}_S+({\mathrm{\&omega;}}_1^2+{\mathrm{\&omega;}}_2^2)P_N}}\hat{S}---\left(18\right)$

By means of similar reason,

With

By scale, that is,

${{\hat{N}}^{\&prime;}}_1=\frac{\sqrt{P_N}}{\sqrt{{({\mathrm{\&omega;}}_3+a{\mathrm{\&omega;}}_4)}^2{P}_S+({\mathrm{\&omega;}}_3^2+{\mathrm{\&omega;}}_4^2)P_N}}{\hat{N}}_1$

${{\hat{N}}^{\&prime;}}_2=\frac{\sqrt{P_N}}{\sqrt{{({\mathrm{\&omega;}}_5+a{\mathrm{\&omega;}}_6)}^2{P}_S+({\mathrm{\&omega;}}_5^2+{\mathrm{\&omega;}}_6^2)P_N}}{\hat{N}}_2---\left(19\right)$

Numerical examples

The normalization power of this direction factor A and S and AS is shown as the function of stereophonic signal level difference and Ф in Fig. 6.

Be used to calculate the weights omega of the least square estimated value of S ₁And ω ₂On Fig. 7, be illustrated as the function of stereophonic signal level difference and Ф in two plates.Be used for

Back scale factor shown in the bottom end plate.

Be used to calculate N ₁Least square estimation and the weights omega of corresponding back scale factor (19) ₃And ω ₂In Fig. 7, be shown the function of stereophonic signal level difference and Ф.

Be used to calculate N ₂Least square estimation and the weights omega of corresponding back scale factor (19) ₅And ω ₆In Fig. 7, be illustrated as the function of stereophonic signal level difference and Ф.

An example that utilizes singer placed in the middle to be used for the stereo rock music folder of spatial decomposition shown in Figure 10.S, A, n ₁And n ₂Estimated value be illustrated.At this signal shown in the time domain, and for each T/F tiling illustrates A.With horizontal sound n independently ₁And n ₂Compare, the direct sound wave s of this estimation is relatively strong, because singer placed in the middle is in the highest flight.

The stereophonic signal of playing decomposition is set in different broadcasts

Provide the spatial decomposition of stereophonic signal, that is, and the direct sound wave of the part that is used to estimate

Direction factor A and horizontal sound independently

With

Sub-band signal, people can define relevant how from different broadcast settings send corresponding to

With

The rule of signal component.

A plurality of loud speakers are in listener's front

Figure 11 illustrates the situation of illustrating.At the virtual sound level width φ shown in the part (a) of this accompanying drawing ₀Virtual sound level width φ ' shown in=30 ° of parts (b) that are scaled at this accompanying drawing ₀, this virtual sound level width φ ' ₀Quilt is by means of a plurality of loudspeaker reproduction.

The independently horizontal sound of this estimation

With

Sent from the loud speaker on this side, for example, the loud speaker 1 and 6 in Figure 11 (b).That is to say that because the horizontal sound that sends from the side is high more, surrounding the listener into, this sound is effective more clearly.Provide the direction factor A of estimation, use " stereo sinusoidal law " (perhaps with other law of the angle of the relevant perception of A) estimation auditory events with respect to ± φ ₀The angle φ of virtual sound level,

$\mathrm{\&phi;}={\sin }^{-1}\left(\frac{A-1}{A+1}\sin {\mathrm{\&phi;}}_0\right)---\left(20\right)$

This angle by convergent-divergent linearly calculating angle with respect to the sound level that enlarges,

${\mathrm{\&phi;}}^{\&prime;}=\frac{{{\mathrm{\&phi;}}^{\&prime;}}_0}{{\mathrm{\&phi;}}_0}\mathrm{\&phi;}---\left(21\right)$

The loud speaker that centers on φ ' is to selected.In the illustrational example of Figure 11 (b), this is to having sign 4 and 5.Be used for this loud speaker between shake the relevant angle γ of amplitude ₀And γ ₁Be defined as in the drawings and illustrate.If the loud speaker of this selection is to having sign 1 and 1+1, so, this signal that these loud speakers provide is:

$a_1\sqrt{1+A^2}S$

$a_2\sqrt{1+A^2}S---\left(22\right)$

Here shake factor a by means of stereo sinusoidal law (perhaps other amplitude is shaken law) calculating and standardization amplitude ₁And a ₂, make $a_1^2+a_2^2=1,$

$a_1=\frac{1}{\sqrt{1+C^2}}$ $a_2=\frac{\mathrm{<figure-callout\; id="1"\; label="C"\; filenames="A20068003222800292.png,A20068003222800301.png"\; state="\{\{state\}\}">C</figure-callout>}}{\sqrt{1+C^2}}---\left(23\right)$

And

$C=\frac{\sin ({\mathrm{\&gamma;}}_0+\mathrm{\&gamma;})}{\sin ({\mathrm{\&gamma;}}_0-\mathrm{\&gamma;})}---\left(24\right)$

The factor in (22)

Be such, the gross power of these signals equals in this stereophonic signal the gross power of relevant component S and AS.Alternatively, people can the use amplitude shake law, and it side by side gives signal to plural loud speaker.

Figure 12 illustrates and is used for loud speaker and goes up the φ ' for M=8 loud speaker to l and l+1 with in angle { 30 ° ,-20 ° ,-12 ° ,-4 °, 4 °, 12 °, 20 °, 30 ° } ₀=φ ₀=30 ° amplitude is shaken factor a ₁And a ₂The example of selecting.

Provide above reason, each T/F tiling of this output signal sound channel shows that i and k are calculated as:

$Y_m=\mathrm{\&delta;}(m-1){{\hat{N}}^{\&prime;}}_1+\mathrm{\&delta;}(m-M){{\hat{N}}^{\&prime;}}_2+(\mathrm{\&delta;}(m-l)a_1+\mathrm{\&delta;}(m-l-1)a_2)\sqrt{1+A^2}{\hat{S}}^{\&prime;}---\left(25\right)$

Here

$\mathrm{\&delta;}\left(m\right)=\left\{\begin{array}{cc}1& \mathrm{for\; m}=0\\ 0& \mathrm{otherwise}\end{array}\right.---\left(26\right)$

And m is output channels sign 1≤m≤M.The sub-band signal of this output channels is converted back to time domain, and forms output channels y ₁To y _MHereinafter, this last step is not always mentioned once more clearly.

The restriction of the scheme of this description is, when the listener is on a side, when for example approaching loud speaker 1, and relatively from the horizontal acoustic phase of opposite side, this laterally independently sound will arrive him with bigger intensity.In order to produce the purpose of two transverse plane ripples, this problem can be evaded by send laterally independently sound from all loud speakers.These are in Figure 13 illustrated.Be somebody's turn to do horizontal independently sound quilt along with the delay of simulating plane wave with certain direction is given to all loud speakers,

$Y_m(i,k)=\frac{{{\hat{N}}^{\&prime;}}_1(i,k-(m-1)d)}{\sqrt{M}}+\frac{{{\hat{N}}^{\&prime;}}_2(i,k-(M-m)d)}{\sqrt{M}}+$

$(\mathrm{\&delta;}(m-l)a_1+\mathrm{\&delta;}(m-l-1)a_2)\sqrt{1+A^2}{\hat{S}}^{\&prime;}---\left(27\right)$

Here d postpones,

$d=\frac{{\mathrm{sf}}_s\sin \mathrm{\&alpha;}}{v}---\left(28\right)$

S is the distance between equally spaced loud speaker, and v is a velocity of sound, f _sBe the sub-band sample frequency, and ± α is two plane directions of wave travel.In our system, this sub-band sample frequency is not sufficiently high, makes d can be expressed as an integer.Therefore, we at first will With

Be converted to time domain, we add its various delay versions to this output channels then.

Loud speaker plus side loud speaker before a plurality of

Previously described broadcast situation purpose is to enlarge virtual sound level, and purpose is to produce the sound level with the perception of listener's location independent.

Optionally, people can utilize the independently horizontal sound of two the independent loudspeaker plays that are arranged at the listener side more With

, as in Figure 14 illustrated.± 30 ° of virtual sound levels (a) are converted to the have loudspeaker array virtual sound level of width in slit of (b).In addition, this laterally independently sound by being play from the side by means of the independent loud speaker that is used for stronger listener's envelope.What people expected is that these results form the stronger impression of listener's envelope.In this case, this output signal is also calculated by (25), and the indication that has sign 1 and M here is the loud speaker on the side.This loud speaker is in this situation selecting l and l+1, makes S ' give never to the signal with index 1 and M, because the whole width of this vitual stage only is projected to preceding loud speaker 2≤m≤M-1.

Figure 15 illustrates an example that is used for being provided with for the identical Musical clip of confession shown in Figure 14 eight signals of generation, and this spatial decomposition that is used for Musical clip is shown in Figure 10.Notice that singer in the highest flight placed in the middle is at two loudspeaker signal y of central authorities ₄And y ₅Between amplitude shake.

5.1 conventional circulating loudspeaker settings

Stereophonic signal is converted to 5.1 possibilities around the multi-channel audio signal of compatibility is to use the setting of having shown in Figure 14 (b) as loud speaker behind three preceding loud speakers arranging with 5.1 standard codes and two.In this case, this back loud speaker sends independently laterally sound, should be used for the reproducing virtual sound level by preceding loud speaker simultaneously.Informally listen to expression and compare with played in stereo, it is more significant when playing the audio signal of describing as listener's envelope.

Stereophonic signal is converted to 5.1 another possibilities around the signal of compatibility is to use as shown in figure 11 setting, this loud speaker is rearranged to mate 5.1 structures here.In this case, ± 30 ° vitual stage be expanded into around the listener ± 110 ° of vitual stages.

Wave field synthesizes Play System

At first, signal y ₁, y ₂... y _MBy with as generation similarly being provided with of Figure 14 (b) illustrated.Then, for each signal y ₁, y ₂... y _M, virtual source is defined in wave field synthesis system.Horizontal sound y independently ₁And y _MBy as sending as plane wave or the information source in the far field illustrational for M=8 among Figure 16.For mutual signal, the virtual source quilt is according to requiring with location definition.In the example shown in Figure 16, this distance changes for different information sources, and some information sources to be defined as be the front of sending array at sound, that is, and can be with special distance this virtual sound level of direction definition for each qualification.

The unitized scheme that is used for 2 to M conversions

Generally speaking, the loudspeaker signal that is used for any description scheme can be illustrated for:

Y＝MN (29)

Here N comprises signal With

Vector.This vector Y comprises all loudspeaker signals.This matrix M has many elements, makes that this loudspeaker signal in vector Y will be identical with what calculated by (25) or (27).As what select, different matrix M can be used filtering and/or different amplitudes to shake law and (for example, use plural loud speaker

Shake) realize.For wave field synthesis system, all loudspeaker signals that this vector Y can comprise this system (normally＞M).In this case, this matrix M also comprises delay, all-pass filter, and filter go usually to realize corresponding to

With

Sending of the wave field of relevant virtual source.In the claims, have delay, all-pass filter and/or be illustrated in the linear combination of element among the N usually as the relational expression of similar (29) of the filter of the matrix element of M.

Revise the audio signal of decomposing

The width of guide sound base

By revising the direction factor of estimation, for example, (i, k), people can control the width of virtual sound level to A.By with greater than 1 factor linear scale direction factor, the instrument that belongs to this sound level is further moved to the side.On the contrary can be by realizing with factor scale less than 1.Alternatively, the people's amplitude that can revise the angle that is used to calculate local direct sound wave is shaken law (20).

Be modified in local direct sound wave and the ratio between the sound independently

In order to control the numerical value of surrounding environment, people can the independently horizontal voice signal of scale

With

So that obtain surrounding environment more or less.Similarly, can utilize scale Signal is revised local direct sound wave aspect intensity.

Revise stereophonic signal

The number that people can also need not to increase sound channel is used to revise the decomposition that stereophonic signal proposes.Here, this purpose only is or revises the width of virtual sound level, perhaps at the direct sound wave of part and the ratio between the sound independently.In this case, the sub-band that is used for this stereo output is:

${\mathrm{<figure-callout\; id="1"\; label="Y"\; filenames="A20068003222800292.png,A20068003222800301.png"\; state="\{\{state\}\}">Y</figure-callout>}}_1=v_1{{\hat{N}}^{\&prime;}}_1+v_2{\hat{S}}^{\&prime;}$ ${\mathrm{<figure-callout\; id="2"\; label="Y"\; filenames="A20068003222800282.png,A20068003222800311.png"\; state="\{\{state\}\}">Y</figure-callout>}}_2=v_1{{\hat{N}}^{\&prime;}}_2+v_2{v}_{3}A{\hat{S}}^{\&prime;}---\left(30\right)$

Here this factor v ₁And v ₂Be used to be controlled at the ratio between sound independently and the local sound.For v ₃≠ 1, same, the width of this sound level is modified (v and in this case, ₂Be modified with compensation for v ₃≠ 1 level variation aspect the sound of part).

Generally turn to plural input sound channel

Show in a word, be used for two input sound channel situations

With Generation following (this is the purpose of lowest mean square estimation).This is sound independently laterally

Be by from X ₁Remove and be included in X equally ₂In signal component calculate.Similarly,

Be by from X ₁Remove and be included in X equally ₁In signal component calculate.Calculate this local direct sound wave

, make it comprise and be present in X ₁And X ₂Signal component among both, and A is the amplitude ratio that calculates, Be comprised in X with this ratio ₁And X ₂In.A represents the direction of local direct sound wave.

As an example, description now has the scheme of four input sound channels.Suppose the loudspeaker signal x that has as in Figure 17 (a) illustrated ₁To x ₄Quadraphonic system be considered to expand to more broadcast sound channel as in Figure 17 (b) illustrated.With similar under two input sound channel situations, calculate independently sound sound channel.In this case, this is four (if perhaps wanting still less) signals

With These signals by with as above for two described identical spirit of input sound channel situation under calculate.That is to say that this is sound independently

Be by from X ₁Remove or be included in equally X ₂Perhaps X ₄Signal component in (signal of adjacent quadrasonics loud speaker) is calculated.Similarly, calculate

With

For each sound channel of adjacent loud speaker to calculating local direct sound wave, that is,

With

Calculate this local direct sound wave

Make it comprise and be present in X ₁And X ₂Signal component among both, and A12 is the amplitude ratio that calculates,

Be included in X with this ratio ₁And X ₂In.A12 represents the direction of local direct sound wave.Because similarly reason is calculated

With

A ₂₃, A ₃₄And A ₄₁In order in the system shown in Figure 17 (b), to play with 12 sound channels, With

By from loud speaker with signal y ₁, y ₄, y ₇And y ₁₂Send.For preceding loud speaker y ₁To y ₄, similar algorithms is applied to for sending

Two input sound channel situations, that is, and the loud speaker of the direction that approaches most to limit by A12 on

Amplitude shake.Similarly, With

Be used as A ₂₃, A ₃₄And A ₄₁Function send from the loudspeaker array that points to three other sides.Alternatively, as under two input sound channel situations, can be used as plane wave and send this independently sound sound channel.Equally, by using for each loud speaker in Figure 17 (b) for the synthetic similar spiritual defining virtual source of the wave field of two input sound channel situations, to play on the wave field synthesis system of listener's loudspeaker array be possible having.Equally, this scheme can be similar to generalization (29), here in this case, vector N comprise all calculating independently with the sub-band signal of the sound sound channel of part.

Because similar reason, 5.1 multichannels can expand to five above main loudspeakers around audio system and play.But center channel needs to note especially because generate content usually here, amplitude shake be applied in left front and right front between (without central authorities).Sometimes amplitude shake also be applied in left front and central between, and between right front and central, perhaps side by side between all three sound channels.Compare with previously described quadrasonics example, this is different, here we used signal imitation supposition only adjacent loud speaker between have public signal component.Therefore perhaps people consider that these remove to calculate local direct sound wave, and perhaps simpler solution is to be mixed into two sound channels under three sound channels with the front, and to use this system description then be quadrasonics.

A kind of be used for will have the scheme expansion of two input sound channels simpler solution that is used for more input sound channel be, some sound channel between use scheme for two input sound channels heuristicly, the synthetic then decomposition that produces with calculated example under quadraphonic situation as

A ₁₂, A ₂₃, A ₃₄And A ₄₁These broadcast can be used as the description for the quadrasonics situation.

Be used for the calculating of ambiophony sound loudspeaker signal

It is irrelevant around audio system that this ambiophony sound system is that characteristics are that signal and specific broadcast are provided with.Single order ambiophony sound system is a characteristic with following signal, and it is defined with respect to some P specific in the space:

W＝S

X＝S?cosψcosФ

Y＝S?sinψcosФ

Z＝S?sin

(31)

Here W=S is (omnidirectional) sound pressure signal in P.This signal X, Y and Z are the signals that obtains from dipole antenna in P, that is, these signals (source point is at a P here) in Cartesian coordinate direction x, y and z are directly proportional with particle rapidity.Angle ψ and Ф represent azimuth and the elevation angle (spherical polar coordinates) respectively.So-called " B form " signal is in addition to be used for W, X, Y and Z

The factor be characteristic.

Be used for M signal on the broadcast system of M accustic channel three-dimension, playing in order to produce, calculate expression from eight direction x ,-x, y ,-y, z ,-signal of the sound that z obtains.This is to finish to obtain directivity (for example, heart-shaped curve) response by synthetic W, X, Y and Z, for example, and (31)

x ₁＝W+X x ₃＝W+Y x ₅＝W+Z

x ₂＝W-X x ₄＝W-Y x ₆＝W-Z

Provide these signals, as being used to calculate eight independently sound sub-band signals (perhaps if desired still less) for describing the similar reason of above quadraphonic system

For example, this sound independently

Be by from X ₁Remove or be included in equally ground, space adjacent channels X ₃, X ₄, X ₅Perhaps X ₆In signal component calculate.In addition, just by adjacent between or the direct sound wave of three times input signal part and the direction factor of representing its direction.Provide this and decompose, as what describe in the quadrasonics example formerly, on loud speaker, send this sound similarly, perhaps common (29).

For the ambiophony sound system of two dimension,

W＝S

X＝S?cos?ψ

Y＝S?sin?ψ (33)

The result forms four input signal x ₁To x ₄, this processing is similar to the quadraphonic system of description.

Matrix ring around decoding

Matrix ring is mixed down stereophonic signal around encoder under with multi-channel audio signal (for example, 5.1 around signal).This form of expression multi-channel audio signal is represented " matrix ring around ".For example, 5.1 can be by matrix encoder with mixing under the following mode (for the sake of simplicity, we ignore the low-frequency effect sound channel) around the sound channel of signal:

$x_1\left(n\right)=l\left(n\right)+\frac{1}{\sqrt{2}}c\left(n\right)+\mathrm{<figure-callout\; id="1"\; label="j"\; filenames="A20068003222800292.png,A20068003222800301.png"\; state="\{\{state\}\}">j</figure-callout>}\frac{1}{\sqrt{2}}{l}_s\left(n\right)+\mathrm{<figure-callout\; id="1"\; label="j"\; filenames="A20068003222800292.png,A20068003222800301.png"\; state="\{\{state\}\}">j</figure-callout>}\frac{1}{\sqrt{6}}{r}_s\left(n\right)$

$x_2\left(n\right)=r\left(n\right)+\frac{1}{\sqrt{2}}c\left(n\right)-\mathrm{<figure-callout\; id="1"\; label="j"\; filenames="A20068003222800292.png,A20068003222800301.png"\; state="\{\{state\}\}">j</figure-callout>}\frac{1}{\sqrt{2}}{r}_s\left(n\right)-\mathrm{<figure-callout\; id="1"\; label="j"\; filenames="A20068003222800292.png,A20068003222800301.png"\; state="\{\{state\}\}">j</figure-callout>}\frac{1}{\sqrt{6}}{l}_s\left(n\right)$

Here I, r, c, l _sAnd r _sRepresent left front, right front, central, a left back and right back sound channel respectively.J represents 90 degree phase shifts, and-j is-90 degree phase shifts.Other matrix encoder can use the modification of the following mixing of description.

With before changed described similarly for 2 to M sound channels, people can be applicable to spatial decomposition that matrix ring is around following mixed frequency signal.Therefore, for each sub-band, sound sub-band independently calculates local sound sub-band and direction factor at every turn.Independently the linear combination of sound sub-band and local sound sub-band is sent by each loud speaker from this surrounding system, that is to say, send the matrix decoding around signal.

Notice and since matrix around the out-phase component in the following mixed frequency signal, the standardized relevant negative value that adopts equally probably.If this situation, corresponding direction factor will be a negative value, be illustrated in the sound channel that sound in the original multi-channel audio signal derives from the back (under matrix before the mixing).

This decoding matrix around mode be very attractive because it has low complexity, and abundant simultaneously surrounding environment is that independently sound sub-band by estimation reproduces.Do not need to produce artificial surrounding environment, it is complete computable aggregate.

The embodiment details

In order to calculate sub-band signal, can use discrete (fast) Fourier transform (DFT).Reduce and the better number of the frequency band that excites of audio quality in order to reduce by complexity, this DFT frequency band can be synthesized and make each synthetic frequency band have the frequency resolution that the frequency resolution by the human auditory system excites.The processing procedure of this description is carried out for each synthetic sub-band then.Alternatively, can use quadrature mirror filter (QMF) group or any other non-cascade or the cascaded filter group.

Two minimum detectable signal types are signals of transient state and static state/tone.In order to illustrate both effectively, can use bank of filters with adaptive T/F resolution mode.With detected transient, and the temporal resolution of this bank of filters (perhaps alternatively, only this processing procedure) will be increased to handle this transient state effectively.Static/the signal component of tone is equally with detected, and the temporal resolution of this bank of filters and/or processing procedure will be lowered for such signal.As the criterion of the signal component that is used to detect stable/tone, people can use " tone measurement ".

Our embodiment of this algorithm uses fast Fourier transform (FFT).For the 44.1kHz sampling rate, we use the FFT size between 256 and 1024.The sub-band that we synthesize has the bandwidth of about human auditory system's twice critical bandwidth.This causes using about 20 synthetic sub-bands for the 44.1kHz sampling rate.

Example application

Television set

In order to play, for the benefit that obtains " stable centers " (for example, the film dialogue appears at the central authorities of screen, is used for all locational listeners) can produce center channel based on stereo audiovisual TV content.Alternatively, if want, stereo can be converted to 5.1 around.

Stereo to the multichannel boxcar

It is a kind of form of playing on plural loud speaker that is applicable to that conversion equipment will be changed audio content.For example, this box can be used to the stereo music player, and is connected to 5.1 speaker units.This user can have multiple choices: the stereo+center channel 5.1 with preceding vitual stage around, with have around the listener ± surrounding environment 5.1 of 110 ° of virtual sound levels around, perhaps all loudspeaker arrangements are used for better/wideer preceding vitual stage in front.

Such boxcar can be input as characteristic with stereo analog line input audio frequency input and/or digital SP-DIF audio frequency.This output or the output of multichannel circuit, perhaps digital audio output alternatively, for example, SP-DIF.

Equipment and device with improved broadcast performance

Around with regard to the audio content, such equipment and device will be supported improved broadcast with comparing traditionally with regard to or multichannel stereo with more loudspeaker plays.In addition, they can to support to change stereo audio content be that multichannel is around content.

The multi-channel loudspeaker device

Multi-channel loudspeaker device prospect has its audio input signal of conversion is used for the signal of each loud speaker for its characteristics performance.

Automobile audio

Automobile audio is a challenge topic.Owing to listener's position with owing to barrier (seat, each listener's human body), and be used for the restriction that loud speaker is placed, it is difficult to play stereo or multi-channel audio signal, makes them reproduce virtual sound level well.The algorithm of this proposition can be used to calculate the signal that is used to be arranged on the loud speaker on the specific position, makes virtual sound level be enhanced for the listener in the central point of club face not.

Other use field

The spatial decomposition that excites with having described the consciousness that is used for stereo and multi-channel audio signal.Laterally independently sound and local sound with and specific angle (perhaps level difference) in many sub-bands and as the function of time, estimated.Provide the signal imitation of a hypothesis, calculate the lowest mean square estimation of these signals.

In addition, how its stereophonic signal of having described this decomposition can be play on a plurality of loud speakers, loudspeaker array and wave field synthesis system.In addition, it has been described the spatial decomposition that proposes and how to be applied to the ambiophony acoustical signal form that " decoding " is used for the multi-channel loudspeaker broadcast.In addition, its outline principle of describing how to be applied to microphone signal, ambiophony sound B format signal and matrix around signal.

Claims (22)

1. One kind from a plurality of input audio tracks (x1 ..., xL) produce a plurality of output audio sound channels (y1 ..., method yM), wherein the number of output channels equals or is higher than the number of input sound channel, the method comprising the steps of:

   -utilize and import sub-band X1 (i) ..., one or more independently sound sub-bands of representing signal component are calculated in the linear combination of XL (i), and this signal component is independently between the input sub-band;

   -utilize and import sub-band X1 (i), ..., the linear combination of XL (i), calculate the direct sound wave sub-band of one or more parts, its expression is comprised in the signal component in the more than one input sub-band, with the corresponding direction factor that calculates the expression ratio, these signal components are included in this ratio in two or more input sub-bands;

   -producing and export sub-band Y1 (i) ... YM (i) comprises step:

   -export sub-band to be set to zero;

   -for each sound sub-band independently, select the subclass of output sub-band, and these are added give the corresponding independently zoom version of sound sub-band;

   -select a pair of output sub-band for each direction factor, and these are added the zoom version of the direct sound wave sub-band of giving corresponding part;

   -will export sub-band, Y1 (i) ... YM (i) is converted to time-domain audio signal y1...yM.
2. 2. according to the method for claim 1, wherein, at least one independently sound sub-band N (i) calculate by remove the signal component that also is present among another input sub-band one or more the sub-band from input, and on a pair of input sub-band of at least one selection

   Local direct sound wave sub-band S (i) calculates according to the signal component that is included in the input sub-band that belongs to accordingly right, and direction factor A (i) is calculated as a ratio, and direct sound wave sub-band S (i) is included in this ratio to be belonged in the corresponding right input sub-band.
3. 3. according to the method for claim 1 or 2, wherein, sound sub-band N (i) independently, local direct sound wave sub-band S (i) and the calculating of direction factor A (i) are calculated as input sub-band X _i(i) ... X _L(i), input sub-band power and the input sub-band between the function of standardization cross-correlation.
4. 4. according to the method for claim 1 to 3, wherein, independently the calculating of the direct sound wave sub-band S (i) of sound sub-band N (i) and part is input sub-band X ₁(i) ... X _L(i) linear combination, the weight of linear combination is here determined by means of the lowest mean square criterion.
5. 5. according to the method for claim 4, wherein, the sub-band power of the independently sound sub-band N (i) of estimation and local direct sound wave sub-band S (i) is adjusted, make its sub-band power equal to be calculated as input sub-band power and import sub-band between the corresponding sub-band power of function of standardized cross-correlation.
6. 6. according to the method for claim 1 to 5, wherein, input sound channel x ₁... x _LOnly be multi-channel audio signal x ₁... x _DThe subclass of sound channel, output channels y here ₁... y _MReplenished by input sound channel with non-processor.
7. 7. according to the process of claim 1 wherein input sound channel x ₁... x _LWith output channels y ₁... y _MCorresponding to the signal that is used to be positioned at respect to the loud speaker on the specific specific direction of listening to the position, and the generation of output signal sub-band is as follows:

   Independently the linear combination of sound sub-band N (i) and local direct sound wave sub-band S (i) makes this output sub-band Y ₁(i) ... Y _M(i) according to following generation:

   Independently sound sub-band N (i) is mixed in the output sub-band, makes the predefined direction of simulation send corresponding sound;

   Local direct sound wave sub-band S (i) is mixed in the output sub-band, makes simulation send corresponding sound by the definite direction of corresponding direction factor A (i).
8. 8. according to the method for claim 7, wherein, sound by sub-band signal being applied to simulate specific direction corresponding to the output sub-band of the loud speaker that approaches specific direction most.
9. 9. according to the method for claim 7, wherein, be applied to simulate specific direction by the identical sub-band signal that will have different gains and sound corresponding to the output sub-band of two loud speakers that directly are adjacent to specific direction.
10. 10. according to the method for claim 7, wherein, be applied to a plurality of output sub-bands by the identical filtering sub-band signal that will have specific delay and gain factor and simulate specific direction with the simulated sound wave field and sound.
11. 11. method according to claim 1 to 10, wherein, this independently sound sub-band N (i), local sound sub-band S (i) and direction factor A (i) be modified attribute with the such width of the virtual sound level of control reproduction, and point to independently sound ratio.
12. 12. according to the method for claim 1 to 11, wherein, the function that all method steps are used as the time repeats.
13. 13. according to the method for claim 12, wherein, the repetition rate of this processing is applicable to specific input signal characteristics, such as, the existence of transient state or static signal component.
14. 14. according to the method for claim 1 to 13, wherein, the number of the criterion chooser frequency band of the frequency resolution of use simulating human auditory system and corresponding sub-band bandwidth.
15. 15. according to any one method of previous claim, wherein, this input sound channel is represented stereophonic signal, and this output channels is represented multi-channel audio signal.
16. 16. according to the method for claim 1 to 14, wherein, the ambient signal of this input stereo audio sound channel representing matrix coding, and this output channels is represented multi-channel audio signal.
17. 17. according to the method for claim 1 to 14, wherein, this input sound channel is a microphone signal, and this output channels is represented multi-channel audio signal.
18. 18. according to the method for claim 1 to 14, wherein, this input sound channel is the linear combination of ambiophony sound B format signal, and this output channels is represented multi-channel audio signal.
19. 19. according to the method for claim 1 to 18, wherein, this output multi-channel audio signal is represented the signal that is used for resetting on wave field synthesis system.
20. 20. an audio conversion devices, wherein this equipment comprises that enforcement of rights requires the device of the step of a method in 1 to 19 the method.
21. 21. according to the audio conversion devices of claim 20, wherein, this equipment is embedded in the audio frequency automotive system.
22. 22. according to the audio conversion devices of claim 20, wherein, this equipment is embedded in TV or the system of cinema.

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